--- channels/chan_sip.c.orig	2008-03-12 17:37:00.000000000 +0200
+++ channels/chan_sip.c	2008-03-12 18:17:33.000000000 +0200
@@ -554,6 +554,9 @@
 static unsigned int global_tos_sip;		/*!< IP type of service for SIP packets */
 static unsigned int global_tos_audio;		/*!< IP type of service for audio RTP packets */
 static unsigned int global_tos_video;		/*!< IP type of service for video RTP packets */
+static int global_force_dtmf_relay = 0;
+static int global_force_dtmf_relay_pt = 101;
+
 static int compactheaders;		/*!< send compact sip headers */
 static int recordhistory;		/*!< Record SIP history. Off by default */
 static int dumphistory;			/*!< Dump history to verbose before destroying SIP dialog */
@@ -4983,6 +4986,8 @@
 	int codec_index = 0;
 	int codec_pt_order[256];
 
+	int dtmf_present = 0;
+
 	if (!p->rtp) {
 		ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
 		return -1;
@@ -5408,12 +5413,20 @@
 	for (x = 0; x < codec_index; ++x) {
 		struct rtpPayloadType pt;
 		pt = ast_rtp_lookup_pt(p->rtp, codec_pt_order[x]);
+		if (pt.code == AST_RTP_DTMF)
+			dtmf_present = 1;
 		if (!pt.isAstFormat && !pt.code && p->vrtp)
 			pt = ast_rtp_lookup_pt(p->vrtp, codec_pt_order[x]);
 		if (pt.isAstFormat)
 			ast_codec_pref_append(&p->formats, pt.code);
 	}
 	ast_codec_pref_remove2(&p->formats, ~p->usercapability);
+	if (!dtmf_present && global_force_dtmf_relay) {
+		newnoncodeccapability |= AST_RTP_DTMF;
+		ast_rtp_set_m_type(newaudiortp, global_force_dtmf_relay_pt);
+		codec_pt_order[codec_index++] = global_force_dtmf_relay_pt;
+		ast_rtp_set_rtpmap_type(newaudiortp, global_force_dtmf_relay_pt, "audio", "telephone-event", 0);
+	}
 
 	/* Now gather all of the codecs that we are asked for: */
 	ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
@@ -16845,6 +16858,9 @@
 
 	global_matchexterniplocally = FALSE;
 
+	global_force_dtmf_relay = 0;
+	global_force_dtmf_relay_pt = 101;
+
 	/* Copy the default jb config over global_jbconf */
 	memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
 
@@ -16901,6 +16917,18 @@
 			}
 		} else if (!strcasecmp(v->name, "vmexten")) {
 			ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
+		} else if (!strcasecmp(v->name, "rtp_force_dtmf_relay")) {
+			if ((global_force_dtmf_relay = ast_true(v->value)))
+				ast_verbose("RTP DTMF relaying will be enforced\n");
+			else
+				ast_verbose("RTP DTMF relaying will not be enforced\n");
+		} else if (!strcasecmp(v->name, "rtp_force_dtmf_relay_pt")) {
+			sscanf(v->value, "%d", &global_force_dtmf_relay_pt);
+			if (global_force_dtmf_relay_pt < 96 || global_force_dtmf_relay_pt > 255) {
+				ast_verbose("RTP forced DTMF relay payload type is not valid: %d. Using default (101)\n", global_force_dtmf_relay_pt);
+				global_force_dtmf_relay_pt = 101;
+			} else
+				ast_log(LOG_WARNING, "RTP forced DTMF relay payload type is %d\n", global_force_dtmf_relay_pt);
 		} else if (!strcasecmp(v->name, "rtptimeout")) {
 			if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
 				ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
--- configs/sip.conf.sample.orig	2008-04-25 08:53:52.000000000 -0700
+++ configs/sip.conf.sample	2008-06-10 00:45:37.000000000 -0700
@@ -53,6 +53,12 @@
 				; and multiline formatted headers for strict
 				; SIP compatibility (defaults to "no")
 
+;rtp_force_dtmf_relay=no	; Enable RFC2833 DTMFs to be sent even if peer
+				; hasn't announced support for it. Default: no
+
+;rtp_force_dtmf_relay_pt=101	; RTP payload type value for enforced RFC2833
+				; DTMFs. Default: 101
+
 ; See doc/ip-tos.txt for a description of these parameters.
 ;tos_sip=cs3                    ; Sets TOS for SIP packets.
 ;tos_audio=ef                   ; Sets TOS for RTP audio packets.